The invention generally relates to long distance communications and more particularly to apparatus and methods for assessing whether or not a link or a path between endpoints in an internet protocol (IP) network is potentially suitable for transmission of telecommunications signals of synchronous signals origin. Examples of sources and destinations of such signals may include, but are not limited to, any of voice, data and image terminal apparatus or any combination thereof.
In telecommunications, time is an essential component in the information content of signals representing audible sounds and in the signal formats of many visibly reproducible signals, such as television signals. By contrast in data communications, preservation of the time component is not nearly so important. The field of telecommunications has long been operated on the basis of circuit switching principles for providing voice communications. In the later half of this century circuit switching networks have carried an ever increasing volume of data communications. The typical telecommunications digital network for communications of voice, digitally encoded in accordance with a pulse code modulation (PCM) standard, provides a continuous bit rate service that is concatenated as nxc3x9764 Kb/s channels. Such telecommunications facilities and networks are said to be circuit switched or synchronous networks, which by their physical natures are most suited to transporting signals between communications terminals which produce synchronous signals, for example telephone speech signals. The primary characteristic of circuit switching is that when one or more physical channels are assigned to a given communication circuit, to provide a service, that channel assignment is reserved for the exclusive use of that service continuously throughout the duration of the service provision. This characteristic of circuit switching is substantially irrelevant for data communications and is of such cost that alternatives, known as packet switched networks, have been developed for the express purpose of providing less costly data communications.
Some time ago, a packet switch, with the trademark SLl was introduced by the assignee, for improving the efficient transport of data signals. In contrast to the steady repetitive nature of PCM signals, data signals for the most part, are bursty or asynchronous in nature. Thus to accommodate the efficient transmission of data signals, a data burst is arranged into a packet of convenient length along with a header which specifies a destination. After a packet has been assembled, a high speed transmission path is allocated, only for a time sufficient to transport the packet of data toward its destination. During the packet transport, the packet is in sole possession of the transmission path. After the packet is transported, the transmission path is available for the transport of another packet, possibly from a different source. The event of transporting at least one packet of data from a point of origin to a point of destination is termed a data call, however, the number of data packets transmitted throughout the duration of a data call is generally unlimited. Packet networks operated in accordance with the internet protocol (IP) have recently become the data communications equivalent of the publicly accessible switched telephone networks (PSTNs). Users of data communications services commonly refer to the xe2x80x9cinternetxe2x80x9d in the same fashion as users of voice communications refer to the xe2x80x9ctelephonexe2x80x9d. If considered on a basis of bulk data transport per unit of cost, the transport of information signals using the IP is very economical as compared with the PSTN. Although digitized voice can be transported via a packet network operated in accordance with the IP, the wide variances of delay caused by the operating characteristics of an IP network tend to deteriorate, distort and occasionally even obliterate the time component. Interrupted, delayed and out of sequenced reception of voice signal packets are common occurrences from time to time in a typical packet system, particularly during higher traffic periods. In other words the IP does not provide a consistent quality of service (QOS) for voice communications and the like.
The general evolution of packet systems toward functionality as broad band carriers of information of synchronous origin is exemplified in a paper by A. Thomas et al, titled Asynchronous Time-Division Techniques: An Experimental Packet Network Integrating Video communication, which was published at the 1984 International Switching Symposium, May 7-11 in Florence Italy. Another example was published in a 1987 IEEE paper by Jean-Pierre Coudreuse and Michel Serval, titled Prelude: An Asynchronous Time-division Switched Network.
More recently, a broadband communications standard for supporting a variety of both synchronous and asynchronous communication requirements has been widely adopted by telecommunications providers, and is now referred to as the asynchronous transfer mode (ATM) of telecommunications. The recommended standards are defined by the ATM Forum and are available from several publishers including Prentice Hall of Englwood Cliffs, N.J. 07632, under the title ATM User-Network Interface Specification Version 3.0 (ISBN 0-13-2258633). One commercially available product is sold by the assignee with the trademark Magellan. Networks operable in the ATM standard are usually termed ATM systems or ATM networks. ATM systems are sophistically compromised to preserve the essence of the time component in synchronous signals yet to some extent reap the economies of packet switching. However, as the IP is strictly directed to the efficient transport of data through any packet network facility, accordingly the IP does not take advantage of the ATM potential for preserving the time component.
The quality of audible speech, reproduced from IP transport, may approach the quality of signals transmitted via the PSTN. Such is usually contingent upon all of the packet network facilities, involved with the transport of the signals, being operated at small fractions,of their capacities. Otherwise the quality may degenerate such that verbal information becomes unintelligible. Never the less, it has become commonplace for some personal computer users to link with an IP network for telephone like voice communications, as well as for data communications. Economies envisaged with utilization of IP networks for synchronous signals communications, have generated considerable development in adaptations of end terminal facilities and software. These adaptations provide degrees of compensation for the irregular delays in packet transport to improve the quality of audibly reproduced speech.
Commercial entities which depend heavily upon telecommunications usage, for their activities, spend significantly upon purchases of telephony services from PSTN and other circuit switched telephone service providers. For some time they have considereded the IP, wishing it were a practical alternative. Improvements in the adaptation of end terminal facilities and software for telephone conversations have made the IP network a potentially practical alternative to the PSTNs. Users depending heavily upon telecommunications find the potential low cost of usage of IP in comparison with the PSTN to be very attractive. Nevertheless, this attraction is tempered with the recognition that from time to time one or more links between the network endpoints related to a telephone call may provide such poor QOS as to be unacceptable. Furthermore the QOS can be unpredictably variable, changing from good to marginal to bad and to good again within an hour.
The effects of packet delays and losses as well as end terminal clocking dissimilarities are discussed in a previous U.S. patent application Ser. No. 08,982,925 assigned to Northern Telecom Ltd., the assignee of this application. The previous application teaches improvements useful in telephone facilities, terminals and personal computers which reduce the potentially deleterious effects of an IP network involved in speech transmission. However, there are practical limits to the effectiveness of these improvements, while there is no limit to the degradation of the performance of an IP link in a packet network.
In the jargon of the IP, a real-time transport protocol has been introduced to distinguish signals of synchronous origin from typical data signals. A signal of synchronous origin is usually referred to as a real-time transport protocol (RTP) stream. This does nothing to expedite the regular transport of these signals, however it has permitted the use of a real-time transport control protocol (RTCP). The RTCP is one of several protocols useful for collecting data relative to characteristics having been inflicted upon RTP streams, while traversing the IP network. Both the RTP and the RTCP are published in standards recommended by the International Telecommunications Union. There are also software tools available which will analyze the collected data and interpret the characteristics of data collected by the RTCP. In other words, such software tool assesses the QOS being momentarily provided via a path of propagation through the IP network. Examples of these software tools are available under the trademarks of V/IP Trunk from Micom, and IP Telecommuter and Road Warrior both from Northern Telecom. Each of these software mechanisms includes functions which take a measure of the performance of an IP connection for synchronous data, such as a telephone call after the call has progressed to a conversation. Of course if the connection provides transport of inadequate quality for real-time voice, the parties to the conversation may very well realize it without the benefit of what is essentially a post performance assessment of the RTP stream. At least one party will likely notice the conversation received from the other party as being delayed, broken or otherwise unsatisfactory. Consequently, as far as telephone voice communications are concerned, many if not most commercial entities are reluctant to becoming committed internet telephony users, as for purposes of their activities, a guaranteed QOS similar to the QOS of the PSTN is virtually essential.
In accordance with the invention, an assessment of a probable QOS for routing an intended telephony call via an IP network is acquired, apriori to actually establishing the telephone call.
In one example, RTCP information, about RTP streams having recently traversed links related to a specific path in the IP network, is gathered to determine, apriori setup of a requested call, a QOS of an IP network path.
More particularly, a quality of service (QOS) server, in combination with an IP network, is responsive to IP path definitions provided thereto from a telecommunications entity coupled with the IP network, for from time to time collecting data relative to characteristics of real-time transport protocol (RTP) streams used for transporting real-time audio data via one of more links in an IP network path.
In one application of the invention a call centre including agent stations, is responsive to data relative to characteristics of real-time transport protocol streams used for transporting real-time audio data via IP paths in an IP network, for selectively postponing an IP telephone call setup with one of the agent stations in an instance wherein the data indicates a probable QOS of less than a predetermined QOS.
In one example the call centre is operative in combination with a quality of service (QOS) server to accommodate outgoing call completions via either of a PSTN and a packet switched network. The call centre comprises a telephone circuit switching network being coupled via trunk circuits to a PSTN, and being coupled via a gateway means to a packet switched network, the telephone circuit switching network being operable to provide communications channels between any of a plurality of agent stations and the gateway means, and to provide communications channels between any of the plurality of agent stations and the trunk circuits. A call controller directs the operations of the telephone circuit switching network for setting up and tearing down telephone calls, such that calls for completion via the trunk circuits are preceded by signalling information specifying at least the telephone number of a called party, and such that calls for completion via the gateway means are preceded by signalling information as to the telephone number of a called party as well as a corresponding IP network endpoint address. An automatic dialler is coupled to communicate with the call controller for providing telephone numbers in association with endpoint addresses when requested by the call controller, and is coupled with the QOS server. The automatic dialler comprises a number list for storing a plurality of telephone numbers along with corresponding endpoint addresses of parties to be called. A calling list controller reads and writes the number list, and apriori to providing a telephone number for use by the call controller, the calling list controller is operable to request and receive QOS information from the QOS server in relation to the endpoint addresses of the call centre""s gateway and an endpoint associated with said telephone number. The calling list controller is further operable to decline provisioning of said telephone number in consequence of a value of said QOS information. Hence in an IP voice call, a predetermined acceptable QOS can be virtually guaranteed.
The automatic dialler in one example includes a deferred list and a called list, in addition to the number list. The deferred list, is for recording telephone numbers for which provisioning was declined. The called list is for storing each telephone number read in the telephone number list and resulting in an IP telephone call setup with a called party having been performed. In this example the calling list controller is responsive to a completion of a reading of all the numbers in the telephone number list for functionally substituting the deferred list; and is responsive to a signal from an agent station during a progress of a telephone connection following the call setup for causing the telephone number of the called party to be recorded in the deferred list.
A method for routing a telephone call via an IP network, is performed in response to a calling party having initiated a call to a called party. The method includes the step of determining at least one IP network link for transport of packets from the calling party to the called party, and at least one IP network link for transport of packets from the called party to the calling party. In accordance with the invention the method comprising the further steps of:
collecting data relative to characteristics of any real-time transport protocol (RTP) streams having recently been transported via said links;
generating a historical quality of service (QOS) value from the collected data; and
setting up the call between the calling and called parties via the IP network, contingent upon the historical QOS value being of at least a predefined QOS;
whereby the setting up an IP network telephony call with a poor QOS is substantially avoided.
The invention provides a method for estimating a telephony quality of service between endpoints in an internet protocol (IP) network. The method requires a data storage facility coupled with the IP network to act as a quality of service (QOS) server. The method is responsive to an exchange of signal streams of synchronous origins at an endpoint and comprises the steps of:
a) gathering performance information for a signal stream received at the end point,
b) gathering route information identifying at least one route having been traversed by said signal stream,
c) transferring the gathered information to the data storage facility; and
in response to each transfer of said gathered information, at the data storage facility,
d) storing the performance information, and the route information;
whereby the stored information are available apriori a call setup for a request of telephone service involving endpoints having at least a potential transport route in common with a route identified in the stored route information.
Following the request for telephone service, the method comprises the further steps of:
e) identifying IP network endpoints of the calling and called parties,
f) transmitting a request for QOS information relative to each of the identified endpoints,
g) responsive to said request, in the QOS server, reading any QOS information for which there is at least a potential transport route in common with a route identified in the stored route information, and
h) transporting any QOS information, read in step d), to the requester,
whereby set up of the requested telephone call via the IP network may be declined if the QOS information appears to indicate less than a prescribed QOS.